VoIP (Voice over Internet Protocol) is one of the popular communication technology. In VoIP, SIP (Session Initiation Protocol) defined by IETF is the most widely used protocol because of its simple structure, expandability and easy operation.
In the present Internet environment, more and more users install NAT (Network Address Translator) servers, but NAT servers induce the communication failure for P2P (Peer to Peer) applications, an SIP server is therefore needed between the NAT servers.
Referring to FIG. 1, which shows the SIP (Session Initiation Protocol) network environment for VoIP, comprises NAT server 1, NAT server 2 and SIP proxy server 3. SIP proxy server 3 is responsible for conducting SIP, i.e. for registration, forwarding or redirection of the computer 4 and computer 5 (client's terminals).
Computer 4 and computer 5 are under NAT server 1 and NAT server 2 respectively, RTP (Real Time Transport Protocol) packets must be transferred through SIP proxy server 3, P2P (peer to peer) communication between Computer 4 and computer 5 is impossible. When a plurality of client's terminals communicates through SIP proxy server 3, it is apparent that the communication efficiency will be reduced significantly.